Reland "Revert "audio_processing/aec: make delay estimator aware of starving farend buffer""

webrtc/modules/audio_processing/aec/aec_core.c

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 839 matching lines @@
   freq_data[1][0] = 0;
   freq_data[1][PART_LEN] = 0;
   freq_data[0][0] = time_data[0];
   freq_data[0][PART_LEN] = time_data[1];
   for (i = 1; i < PART_LEN; i++) {
     freq_data[0][i] = time_data[2 * i];
     freq_data[1][i] = time_data[2 * i + 1];
   }
 }
 
+static int MoveFarReadPtrWithoutSystemDelayUpdate(AecCore* self, int elements) {
+  WebRtc_MoveReadPtr(self->far_buf_windowed, elements);
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+  WebRtc_MoveReadPtr(self->far_time_buf, elements);
+#endif
+  return WebRtc_MoveReadPtr(self->far_buf, elements);
+}
+
 static int SignalBasedDelayCorrection(AecCore* self) {
   int delay_correction = 0;
   int last_delay = -2;
   assert(self != NULL);
 #if !defined(WEBRTC_ANDROID)
   // On desktops, turn on correction after |kDelayCorrectionStart| frames.  This
   // is to let the delay estimation get a chance to converge.  Also, if the
   // playout audio volume is low (or even muted) the delay estimation can return
   // a very large delay, which will break the AEC if it is applied.
   if (self->frame_count < kDelayCorrectionStart) {
@@ skipping 20 matching lines @@
     int delay = last_delay - WebRtc_lookahead(self->delay_estimator);
     // Allow for a slack in the actual delay, defined by a |lower_bound| and an
     // |upper_bound|.  The adaptive echo cancellation filter is currently
     // |num_partitions| (of 64 samples) long.  If the delay estimate is negative
     // or at least 3/4 of the filter length we open up for correction.
     const int lower_bound = 0;
     const int upper_bound = self->num_partitions * 3 / 4;
     const int do_correction = delay <= lower_bound || delay > upper_bound;
     if (do_correction == 1) {
       int available_read = (int)WebRtc_available_read(self->far_buf);
-      // Adjust w.r.t. a |shift_offset| to account for not as reliable estimates
-      // in the beginning, hence we are more conservative.
-      delay_correction = -(delay - self->shift_offset);
+      // With |shift_offset| we gradually rely on the delay estimates.  For
+      // positive delays we reduce the correction by |shift_offset| to lower the
+      // risk of pushing the AEC into a non causal state.  For negative delays
+      // we rely on the values up to a rounding error, hence compensate by 1
+      // element to make sure to push the delay into the causal region.
+      delay_correction = -delay;
+      delay_correction += delay > self->shift_offset ? self->shift_offset : 1;
       self->shift_offset--;
       self->shift_offset = (self->shift_offset <= 1 ? 1 : self->shift_offset);
       if (delay_correction > available_read - self->mult - 1) {
         // There is not enough data in the buffer to perform this shift.  Hence,
         // we do not rely on the delay estimate and do nothing.
         delay_correction = 0;
       } else {
         self->previous_delay = last_delay;
         ++self->delay_correction_count;
       }
@@ skipping 520 matching lines @@
   }
   // We create the delay_estimator with the same amount of maximum lookahead as
   // the delay history size (kHistorySizeBlocks) for symmetry reasons.
   aec->delay_estimator = WebRtc_CreateDelayEstimator(
       aec->delay_estimator_farend, kHistorySizeBlocks);
   if (aec->delay_estimator == NULL) {
     WebRtcAec_FreeAec(aec);
     return NULL;
   }
 #ifdef WEBRTC_ANDROID
+  aec->reported_delay_enabled = 0;  // DA-AEC enabled by default.
   // DA-AEC assumes the system is causal from the beginning and will self adjust
   // the lookahead when shifting is required.
   WebRtc_set_lookahead(aec->delay_estimator, 0);
 #else
+  aec->reported_delay_enabled = 1;
   WebRtc_set_lookahead(aec->delay_estimator, kLookaheadBlocks);
 #endif
+  aec->extended_filter_enabled = 0;
 
   // Assembly optimization
   WebRtcAec_FilterFar = FilterFar;
   WebRtcAec_ScaleErrorSignal = ScaleErrorSignal;
   WebRtcAec_FilterAdaptation = FilterAdaptation;
   WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress;
   WebRtcAec_ComfortNoise = ComfortNoise;
   WebRtcAec_SubbandCoherence = SubbandCoherence;
 
 #if defined(WEBRTC_ARCH_X86_FAMILY)
@@ skipping 126 matching lines @@
   aec->delay_median = -1;
   aec->delay_std = -1;
   aec->fraction_poor_delays = -1.0f;
 
   aec->signal_delay_correction = 0;
   aec->previous_delay = -2;  // (-2): Uninitialized.
   aec->delay_correction_count = 0;
   aec->shift_offset = kInitialShiftOffset;
   aec->delay_quality_threshold = kDelayQualityThresholdMin;
 
-#ifdef WEBRTC_ANDROID
-  aec->reported_delay_enabled = 0;  // Disabled by default.
-#else
-  aec->reported_delay_enabled = 1;
-#endif
-  aec->extended_filter_enabled = 0;
   aec->num_partitions = kNormalNumPartitions;
 
   // Update the delay estimator with filter length.  We use half the
   // |num_partitions| to take the echo path into account.  In practice we say
   // that the echo has a duration of maximum half |num_partitions|, which is not
   // true, but serves as a crude measure.
   WebRtc_set_allowed_offset(aec->delay_estimator, aec->num_partitions / 2);
   // TODO(bjornv): I currently hard coded the enable.  Once we've established
   // that AECM has no performance regression, robust_validation will be enabled
   // all the time and the APIs to turn it on/off will be removed.  Hence, remove
@@ skipping 97 matching lines @@
   TimeToFrequency(fft, xf, 0);
   WebRtc_WriteBuffer(aec->far_buf, &xf[0][0], 1);
 
   // Convert far-end partition to the frequency domain with windowing.
   memcpy(fft, farend, sizeof(float) * PART_LEN2);
   TimeToFrequency(fft, xf, 1);
   WebRtc_WriteBuffer(aec->far_buf_windowed, &xf[0][0], 1);
 }
 
 int WebRtcAec_MoveFarReadPtr(AecCore* aec, int elements) {
-  int elements_moved = WebRtc_MoveReadPtr(aec->far_buf_windowed, elements);
-  WebRtc_MoveReadPtr(aec->far_buf, elements);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  WebRtc_MoveReadPtr(aec->far_time_buf, elements);
-#endif
+  int elements_moved = MoveFarReadPtrWithoutSystemDelayUpdate(aec, elements);
   aec->system_delay -= elements_moved * PART_LEN;
   return elements_moved;
 }
 
 void WebRtcAec_ProcessFrames(AecCore* aec,
                              const float* const* nearend,
                              int num_bands,
                              int num_samples,
                              int knownDelay,
                              float* const* out) {
@@ skipping 52 matching lines @@
       // 2 a) Compensate for a possible change in the system delay.
 
       // TODO(bjornv): Investigate how we should round the delay difference;
       // right now we know that incoming |knownDelay| is underestimated when
       // it's less than |aec->knownDelay|. We therefore, round (-32) in that
       // direction. In the other direction, we don't have this situation, but
       // might flush one partition too little. This can cause non-causality,
       // which should be investigated. Maybe, allow for a non-symmetric
       // rounding, like -16.
       int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
-      int moved_elements = WebRtc_MoveReadPtr(aec->far_buf, move_elements);
-      WebRtc_MoveReadPtr(aec->far_buf_windowed, move_elements);
+      int moved_elements =
+          MoveFarReadPtrWithoutSystemDelayUpdate(aec, move_elements);
       aec->knownDelay -= moved_elements * PART_LEN;
-  #ifdef WEBRTC_AEC_DEBUG_DUMP
-      WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
-  #endif
     } else {
       // 2 b) Apply signal based delay correction.
       int move_elements = SignalBasedDelayCorrection(aec);
-      int moved_elements = WebRtc_MoveReadPtr(aec->far_buf, move_elements);
-      WebRtc_MoveReadPtr(aec->far_buf_windowed, move_elements);
-  #ifdef WEBRTC_AEC_DEBUG_DUMP
-      WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
-  #endif
+      int moved_elements =
+          MoveFarReadPtrWithoutSystemDelayUpdate(aec, move_elements);
+      int far_near_buffer_diff = WebRtc_available_read(aec->far_buf) -
+          WebRtc_available_read(aec->nearFrBuf) / PART_LEN;
       WebRtc_SoftResetDelayEstimator(aec->delay_estimator, moved_elements);
       WebRtc_SoftResetDelayEstimatorFarend(aec->delay_estimator_farend,
                                            moved_elements);
       aec->signal_delay_correction += moved_elements;
-      // TODO(bjornv): Investigate if this is reasonable.  I had to add this
-      // guard when the signal based delay correction replaces the system based
-      // one. Otherwise there was a buffer underrun in the "qa-new/01/"
-      // recording when adding 44 ms extra delay.  This was not seen if we kept
-      // both delay correction algorithms running in parallel.
-      // A first investigation showed that we have a drift in this case that
-      // causes the buffer underrun.  Compared to when delay correction was
-      // turned off, we get buffer underrun as well which was triggered in 1)
-      // above.  In addition there was a shift in |knownDelay| later increasing
-      // the buffer.  When running in parallel, this if statement was not
-      // triggered.  This suggests two alternatives; (a) use both algorithms, or
-      // (b) allow for smaller delay corrections when we operate close to the
-      // buffer limit.  At the time of testing we required a change of 6 blocks,
-      // but could change it to, e.g., 2 blocks. It requires some testing
-      // though.
-      if ((int)WebRtc_available_read(aec->far_buf) < (aec->mult + 1)) {
-        // We don't have enough data so we stuff the far-end buffers.
-        WebRtcAec_MoveFarReadPtr(aec, -(aec->mult + 1));
+      // If we rely on reported system delay values only, a buffer underrun here
+      // can never occur since we've taken care of that in 1) above.  Here, we
+      // apply signal based delay correction and can therefore end up with
+      // buffer underruns since the delay estimation can be wrong.  We therefore
+      // stuff the buffer with enough elements if needed.
+      if (far_near_buffer_diff < 0) {
+        WebRtcAec_MoveFarReadPtr(aec, far_near_buffer_diff);
       }
     }
 
     // 4) Process as many blocks as possible.
     while (WebRtc_available_read(aec->nearFrBuf) >= PART_LEN) {
       ProcessBlock(aec);
     }
 
     // 5) Update system delay with respect to the entire frame.
     aec->system_delay -= FRAME_LEN;
@@ skipping 93 matching lines @@
 int WebRtcAec_extended_filter_enabled(AecCore* self) {
   return self->extended_filter_enabled;
 }
 
 int WebRtcAec_system_delay(AecCore* self) { return self->system_delay; }
 
 void WebRtcAec_SetSystemDelay(AecCore* self, int delay) {
   assert(delay >= 0);
   self->system_delay = delay;
 }